Delta-sigma ADC question

[Al Clark]

Additionally, you might have to deal with delays. Typical delay times
between analog input and digital output of delta-sigma ADs are around 15
samples! This might make synching mux timing and sampled data hard to do.

This delay is exactly why SAR converters are used in feedback systems.

A very big advantage to sigma delta converters is that the analog
antialiasing filters are trivial since the actual sample rate is much
high than the "effective" sample rate.

The internal digital filters may have stopband attenuation of 90dB at
something near 1/2 the "sample rate" This of course is a major reason for
the added delay. The better converters have better filters which have
more delay.

You may find that low cost sigma delta converters are still a better
choice than a muxed SAR. You can find 4 channel devices that smaple at
100kHz or 200kHz for reasonable cost.

 

[Lasse Langwadt Christensen]

I've never understood that. You'll have to explain it to me some day.

John

the simple way:

If you model the quantization as as adding white noise with a power
proportional to the number of bits, you'll see that for a delta-sigma
modulator that noise will show up at the output high-passed, the power
is still the same but "pushed" towards higher frequencies.
When you then lowpass and decimate you remove the higher frequencies
and thus much of the the noise.

quatization noise before lowpass and decimate:

^
| - -
| /
| /
| /
+==------+> f
fs/2
quatization noise after lowpass and decimate:

^
|
|
|
|
+==+------> f
fs/2

With a SAR the noise is flat at the output, so lowpass and decimate
will remove less of the noise power.


quatization noise before lowpass and decimate:

^
|
|
|
|---------
+---------+>
fs/2

quatization noise after lowpass and decimate:

^
|
|
|
|--
+--+-----> f
fs/2


-Lasse

 

[Jon Harris]

This delay is exactly why SAR converters are used in feedback systems.

A very big advantage to sigma delta converters is that the analog
antialiasing filters are trivial since the actual sample rate is much
high than the "effective" sample rate.

The internal digital filters may have stopband attenuation of 90dB at
something near 1/2 the "sample rate" This of course is a major reason for
the added delay. The better converters have better filters which have
more delay.

I have noticed a trend recently where some of the higher-end audio converters
(especially DACs) are offering lower latency outputs, either standard, or as an
option. Before, no one seemed to care too much or talk about it, but the
manufacturers are starting to make a bigger deal about latency/group delay.
Just thought I'd mention this.